To aid understanding of the central idea on which the invention is based, there follows below a brief explanation of the main aspects and capabilities, advantages and disadvantages, of packet-switched data transmission, which is used in the field of VoIP-applications, and also of the main embodiments of internet telephony according to the prior art.
In packet-switched networks, a type of transmission is implemented in which no continuous, physical channel is made available for a connection. Instead, the message to be transmitted is divided at the transmission end into small message packets—which are sometimes of varying lengths—which are provided with additional data for identifying the transmitter and receiver and are continuously numbered in the header. These packets are then sent through the network individually, independently of one another. Each packet is analyzed in the individual network nodes. Depending on the current network load, a decision is then made in the individual network nodes for each packet to determine via which outgoing connection the packet in question is to be forwarded to the receiver.
Consequently the packets sometimes take different routes to the receiver depending on the network utilization and are not received in the order in which they are sent. For this reason the packets sometimes need to be buffered in the individual network nodes and sorted in the receiver.
The advantages of packet-switching technology lie in good network utilization and the facilitation of communication between terminals that have different connection rates. However, packet-switched networks are rarely used for time-critical applications since a lot of time is required for the processing of the packets in the individual network nodes, the transmission paths of varying length, and the assembly and disassembly of the packets.
The terms “internet telephony”, “IP telephony” and “voice-over IP” (VoIP) are used to describe a type of communication for transmitting voice data between two communication parties via the internet and/or an intranet, in which computers or conventional fixed-network, cordless, or mobile telephones are used as communication terminals. In addition to voice transmission and e-mail, fax and video services as well as voice mail are possible via the internet and/or intranet. The standardization of hardware and software for these various forms of communication is often referred to in the literature by the term “unified messaging”.
In internet telephony, two originally separate types of network are usually involved in the telephone network and the internet. Conventional telephone connections are normally used for the sections of the path from the calling or called communication party to the nearest network dial-in node. On the other hand, the internet is used for the sections of the path between the relevant network dial-in nodes, which are usually very much longer.
Various connection scenarios can result depending on which technology is used by the parties involved in an internet telephone call. The communication flows either between different computers, between a computer and a normal telephone, or between different telephones. This essentially means that there are four different variants of internet telephony:
Variant 1: Computer⇄Internet⇄Computer
In this variant, the user dials into the internet from his or her computer, e.g. a PC, via a “provider”. The user then tries to dial the fixed IP address of the required communication party via his or her telephony software. If the computer of the required communication party is connected to the internet and that party has loaded his or her telephony software, he or she may receive the call. In this case the computers of the two communication parties are connected via the PSTN (public switched telephone network) to an ITSP (internet telephony service provider), from where they can transmit voice data in the packet-switched internet.
Variant 2: Computer⇄Internet⇄Telephone
In this variant the user dials into the internet from his or her computer via his or her provider. He or she then dials the number of the required communication party using the telephony software. The data packets are sent by the software to a gateway that is nearest geographically to the required communication party. Gateways are special interfaces between networks of different network operators or national networks, which link together private branch exchanges or switching centers via the internet and are used for recording call charges, converting different signaling procedures, and for speed adaptation. With the help of gateways, it is possible—for example—to telephone from one conventional telephone connection to another conventional telephone connection by dialing the telephone number and using a special internet access code. The voice data is transferred to the local PSTN from the nearest gateway. This method places great demands on the supplier's infrastructure. To enable internet telephony to be provided cost-effectively, at least one gateway between the internet and the local telephone network must be installed in every country in the world.
Variant 3: Telephone⇄Internet⇄Computer
To reach a telephone via a computer, it is necessary to set up a connection to an internet telephone service provider. Only then can the required communication party be dialed. For this procedure to work, it is necessary for the computer of the required communication party to be switched on, for their telephony software to be loaded and for their computer to be connected to the internet.
Variant 4: Telephone⇄Internet⇄Telephone
In this solution the user dials into a gateway from his or her telephone connection via the circuit-switched PSTN. After dialing into a gateway the user dials the internet access code or PIN “Personal Identification Number” allocated to him or her by the network operator, and finally the destination call number of the required communication party. From the dialed destination call number or part thereof, the gateway system then determines which gateway is geographically nearest to this destination using a routing table. The gateway—for its part—is identifiable by an IP address, and requires this in order to implement a connection to the required communication party via their local PSTN. As soon as this happens, the calling communication party receives a signal and can speak. The voice data is transmitted via the internet using packet-switching technology.
All four variants can be implemented either via the worldwide internet or via a corporate or organizational intranet, with intranets having better transmission quality as a rule.
A fundamental element of an internet telephony system is the call processing server (CPS), often also known as the gatekeeper. This consists of a series of software applications that run on one or more servers. These may be located at any point within the logical IP network. In its simplest form, a CPS provides an overview of the status of all clients belonging to a certain domain. Its functions are defined according to the H.323 standard of the International Telecommunications Union (ITU). These functions include the resolution of addresses (from E.164 to IP and vice versa) and various authentication and authorization tasks, central call processing, and routing. In addition, it conducts switching functions (call control including call setup and call release) for clients and gateways within the IP network and manages a database in which user profile and network configuration information is stored. The functionality provided by the CPS does, however, vary greatly from one manufacturer to another.
If individual voice channels cannot be switched via the internet and alternative connections to the voice-over IP connection must be switched via circuit-switched networks, the great variety of features of the VoIP connection that can be controlled by the CPS are sometimes lost to the user. In this case, certain features—such as, for example, the setting up of conference calls—can no longer be used.
An essential characteristic of circuit-switched and/or packet-switched data traffic in the ISDN is the consistent separation between the transmission of signaling data and useful data, which is carried out on different channels. Of primary importance in this are the so-called B and D channels. A first B-channel with a data transfer rate of 64 kbps is used for the transfer of digitized voice signals. In parallel to this the user may be offered a second B-channel for transmitting data, which likewise has a data transfer rate of 64 kbps. At the same time a complete S0 interface is provided which permits up to eight different communication terminals to be connected for each user, even if a data connection is not expected to be present at the telephone of a user. This means that both B-channels are always available in both communication directions and that several communication terminals can be active at the same time, each of them using one of the two B-channels simultaneously. In contrast, a D-channel with a data transfer rate of 16 kbps (D16) or 64 kbps (D64) is used in addition for the transfer of signaling data. Besides the exchange of signaling data, users can also send data packets to the network on the D-channel, and these are forwarded in turn by the network to other communication parties. However, unlike in the case of the B-channel, connections cannot be set up via the D-channel.
According to the prior art, the strict separation of signaling and useful data guarantees that the user does not have to suffer any loss of information or features, even if the useful data is diverted via an alternative network, since the D-channel always transfers the signaling and control information correctly.